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<font size="-1"><font face="Helvetica, Arial, sans-serif">With tcpdump
I saw that there were packets coming in from the GSM-gateway to the
public Asterisk-server.<br>
I saw nothing on the Asterisk-CLI that told me that there were attempts
to register, but a "sip debug" shows this :<br>
<br>
<small><font color="#333399">&lt;------------&gt;<br>
[Apr 20 15:07:41] Scheduling destruction of SIP dialog
'<a class="moz-txt-link-abbreviated" href="mailto:0cd637c143e6667c4b5279b713b50403@192.168.1.25">0cd637c143e6667c4b5279b713b50403@192.168.1.25</a>' in 32000 ms (Method:
REGISTER)<br>
[Apr 20 15:07:41] Really destroying SIP dialog
'<a class="moz-txt-link-abbreviated" href="mailto:5c4fc9a47a3f5d545608747f451869f9@192.168.1.25">5c4fc9a47a3f5d545608747f451869f9@192.168.1.25</a>' Method: REGISTER<br>
[Apr 20 15:07:41] <br>
&lt;--- SIP read from my_public_ip:5066 ---&gt;<br>
REGISTER <a class="moz-txt-link-freetext" href="sip:my_asterisk_ip">sip:my_asterisk_ip</a> SIP/2.0<br>
Via: SIP/2.0/UDP 192.168.1.25:5066;rport;branch=z9hG4bK959181a47a<br>
From: "SIM 3" <a class="moz-txt-link-rfc2396E" href="sip:simsim3@my_asterisk_ip">&lt;sip:simsim3@my_asterisk_ip&gt;</a>;tag=4c9ddc99<br>
To: "SIM 3" <a class="moz-txt-link-rfc2396E" href="sip:simsim3@my_asterisk_ip">&lt;sip:simsim3@my_asterisk_ip&gt;</a><br>
Call-ID: <a class="moz-txt-link-abbreviated" href="mailto:001816f82d8bf3d43f34faa82f344c03@192.168.1.25">001816f82d8bf3d43f34faa82f344c03@192.168.1.25</a><br>
Contact: <a class="moz-txt-link-rfc2396E" href="sip:simsim3@192.168.33.104:5060">&lt;sip:simsim3@192.168.33.104:5060&gt;</a><br>
CSeq: 2354 REGISTER<br>
Max-Forwards: 70<br>
Expires: 60<br>
Allow: INVITE,CANCEL,ACK,BYE,NOTIFY,REFER,OPTIONS,INFO,MESSAGE,UPDATE<br>
User-Agent: Mv-37x (904290)<br>
Content-Length: 0<br>
<br>
<br>
&lt;-------------&gt;<br>
[Apr 20 15:07:41] --- (12 headers 0 lines) ---<br>
[Apr 20 15:07:41] Using latest REGISTER request as basis request<br>
[Apr 20 15:07:41] Sending to my_public_ip : 5066 (NAT)<br>
[Apr 20 15:07:41] <br>
&lt;--- Transmitting (NAT) to my_public_ip:5066 ---&gt;<br>
SIP/2.0 100 Trying<br>
Via: SIP/2.0/UDP
192.168.1.25:5066;branch=z9hG4bK959181a47a;received=my_public_ip;rport=5066<br>
From: "SIM 3" <a class="moz-txt-link-rfc2396E" href="sip:simsim3@my_asterisk_ip">&lt;sip:simsim3@my_asterisk_ip&gt;</a>;tag=4c9ddc99<br>
To: "SIM 3" <a class="moz-txt-link-rfc2396E" href="sip:simsim3@my_asterisk_ip">&lt;sip:simsim3@my_asterisk_ip&gt;</a><br>
Call-ID: <a class="moz-txt-link-abbreviated" href="mailto:001816f82d8bf3d43f34faa82f344c03@192.168.1.25">001816f82d8bf3d43f34faa82f344c03@192.168.1.25</a><br>
CSeq: 2354 REGISTER<br>
User-Agent: Asterisk PBX<br>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY<br>
Supported: replaces<br>
Content-Length: 0<br>
<br>
<br>
&lt;------------&gt;<br>
[Apr 20 15:07:41] <br>
&lt;--- Transmitting (NAT) to my_public_ip:5066 ---&gt;<br>
SIP/2.0 401 Unauthorized<br>
Via: SIP/2.0/UDP
192.168.1.25:5066;branch=z9hG4bK959181a47a;received=my_public_ip;rport=5066<br>
From: "SIM 3" <a class="moz-txt-link-rfc2396E" href="sip:simsim3@my_asterisk_ip">&lt;sip:simsim3@my_asterisk_ip&gt;</a>;tag=4c9ddc99<br>
To: "SIM 3" <a class="moz-txt-link-rfc2396E" href="sip:simsim3@my_asterisk_ip">&lt;sip:simsim3@my_asterisk_ip&gt;</a>;tag=as09b99e8c<br>
Call-ID: <a class="moz-txt-link-abbreviated" href="mailto:001816f82d8bf3d43f34faa82f344c03@192.168.1.25">001816f82d8bf3d43f34faa82f344c03@192.168.1.25</a><br>
CSeq: 2354 REGISTER<br>
User-Agent: Asterisk PBX<br>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY<br>
Supported: replaces<br>
WWW-Authenticate: Digest algorithm=MD5, realm="103001vc",
nonce="3c911c4a"<br>
Content-Length: 0</font></small><br>
<br>
How come there is a register attempt that is "Unauthorized" and how
come this doesn't show on the CLI ??<br>
<br>
<br>
Kind regards,<br>
<br>
Jonas.<br>
<br>
</font></font><br>
Jonas Kellens wrote:
<blockquote cite="mid:4BCD78B4.5010407@telenet.be" type="cite"><font
 size="-1"><font face="Helvetica, Arial, sans-serif">Hello list,<br>
  <br>
has anyone experience with the Portech MV-374 GSM-gateway ?<br>
  <br>
I'm trying to register the SIP-accounts to a public SIP-server but that
fails.<br>
  <br>
When trying to register to a local Asterisk-server, all goes well.<br>
  <br>
So anyone knows what special setting I need to make to register my
SIP-accounts/SIM-cards to a public IP ??</font></font></blockquote>
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